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• FFMpeg/FFServer P2P streaming between embedded device and smartphone

  25 avril 2016, par Bogdan Emil Mariesan

  I’ve been playing for the last couple of days with FFMpeg and FFServer as I am considering them as candidates for livestreaming between an embedded device(with an integrated HD camera) and various clients(smartphones).

  I’ve managed to achieve the stream using with the following config for FFServer :

  HTTPPort 1234
  
  RTSPPort 1235
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 512000             # Maximum bandwidth per client
  
                                 # set this high enough to exceed stream bitrate
  
  CustomLog -
  
  NoDaemon                       # Remove this if you want FFserver to daemonize after start
  
  <feed>               # This is the input feed where FFmpeg will send
  
    File /tmp/feed.ffm            # video stream.
  
    FileMaxSize 512K
  
  </feed>
  
  <stream>                      # Output stream URL definition
  
     Feed feed.ffm              # Feed from which to receive video
  
     Format rtp
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 720x576           # Video resolution
  
     VideoFrameRate 60           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
     AVOptionVideo cpu-used 0
  
     AVOptionVideo qmin 10
  
     AVOptionVideo qmax 42
  
     AVOptionVideo quality good
  
     AVOptionAudio flags +global_header
  
     PreRoll 15
  
     StartSendOnKey
  
     VideoBitRate 400            # Video bitrate
  
     NoAudio
  
  </stream>

  And the following FFMpeg command to send the stream to FFServer :

  ffmpeg -rtbufsize 2100M -f dshow -i video="Integrated Camera" -vcodec libx264 http://127.0.0.1:1234/feed.ffm

  I also have a simple Android client that plays the RTSP stream using the following URL :

  rtsp://mylocalnetworkip:1235/test.h264

  But now I am trying to achieve a P2P connection between the embedded device and a smartphone. This has to be over the internet(not in the LAN) and capable to achieve UDP hole punching (such as Skype does for p2p video-calling).

  1. Is this achievable with ffmpeg alone ?
  2. Can ffmpeg integrate with a Stun/Turn server such as Coturn to bypass symmetric NATs ?

• Opencv/FFMPEG retrieving wrong fps while streaming cameras

  15 avril 2016, par Romanzo Criminale

  I’ve have been trying to retrieve the fps information from different camera stream via RTSP. This is done using the OpenCV library which uses the libavormat provided by ffmpeg. However it seems I can’t rely on the information returned by libav depending on the codec or the camera used. I would like to get a better understanding about how this frame rate are computed etc...

  OpenCV is trying to retrieve the fps of the stream using different properties of AVStream, r_frame_rate, avg_frame_rate and time_base.

  double CvCapture_FFMPEG::get_fps() const
  
  {
  
   #if LIBAVCODEC_BUILD >= CALC_FFMPEG_VERSION(54, 1, 0)
  
    double fps = r2d(ic->streams[video_stream]->avg_frame_rate);
  
   #else
  
    double fps = r2d(ic->streams[video_stream]->r_frame_rate);
  
   #endif
  
  
  
   #if LIBAVFORMAT_BUILD >= CALC_FFMPEG_VERSION(52, 111, 0)
  
    if (fps < eps_zero)
  
    {
  
      fps = r2d(ic->streams[video_stream]->avg_frame_rate);
  
    }
  
   #endif
  
  
  
   if (fps < eps_zero)
  
   {
  
    fps = 1.0 / r2d(ic->streams[video_stream]->codec->time_base);
  
   }
  
  
  
   return fps;
  
  }

  I tried two different camera brands and two different codecs H264, and JPEG. Here are the results, I get :

  Camera 1 running at 15fps (default value)

  camera 1, jpeg

  • r_frame_rate = 15/1
  • avg_frame_rate = 0/0
  • time_base = 1/ 90000
  • tick_per_frame = 1

  camera 1, h264

  • r_frame_rate = 15/1

  • avg_frame_rate = 15/2

  • time_base = 1/ 90000

  • tick_per_frame = 2

  Camera 2 running at 25fps default but the fps is changed within the rtsp url to 12

  camera 2, jpeg

  • r_frame_rate = 12/1

  • avg_frame_rate = 0/0

  • time_base = 1/ 90000

  • tick_per_frame = 1

  camera 2, h264 running at 12fps

  • r_frame_rate = 25/1

  • avg_frame_rate = 0/0

  • time_base = 1/ 90000

  • tick_per_frame = 2

  Also i tried streaming using ffmpeg command line

  ffmpeg -i rtsp://url

  and it seems that only jpeg can return the expected fps.

  Here are the questions have been asking myself :

  1/ what’s really r_frame_rate ?
  It says in the api "[i]it is the smallest frame rate that can accurately represent all timestamps. For example, if you have mixed 25 and 30 fps content, then
  will be 150 (it is the least common multiple).[/i]" I doesn’t make sense to me that it the same time the smallest frame rate (25) and the least common multiple (150).

  2/ How is the avg_frame_rate computed ?
  I had a look at the code but couldn’t really figure out how is this average computed. It seems that most of the time this is returning 0.

  3/ Is there a bug getting the frame rate while using h264 ?
  Using jpeg, it looks like r_frame_RATE can return the actual fps value, but h264 will not work. Are those information not available using h264 ?

  Any help would be appreciated, I spent two days trying to understand why i wasn’t getting the correct values but at this point i’m pretty stuck.
  Thanks in advance !

• FFMpeg mux mp4 format result in streams not in file

  5 mars 2016, par M.Taha

  I am new to ffmpeg, i have spend more than 10 days on finding any way to do muxing in mp4 format with audio and vedio in streams buffer not in file.
  What i want is to mux mp4 format audio & vedio in a streams.

  I am able to do muxing mp4 format in file. But not able to get mux mp4 format in streams buffer.

  Till now i have tried this :
  avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, 1, &bd, NULL, &write_packet, NULL) ;
  By calling this avio_alloc_context and passing reference of write_packet function. I am able to get call write_packet. But when i write the data coming in write_packet in a file, and making mp4 file with that. The resultant mp4 file is not working. There is no vedio or audio information available by watching this file in Media Info.
  The header is written , then loop is written and finally trailer is written, but not getting success in running final file.

  Is there any good way to do mux in mp4 format in streams, so please tell me.

  Kindly help to me to do this.

  Thanks in advance.